Bluetooth is an international open standard that allows devices to wirelessly communicate with each other. Bluetooth is a short-range wireless technology that allows Bluetooth enabled devices such as computers, cell phones, keyboards and headphones to establish connections without using wires or cables to couple the devices to each other. Bluetooth is currently incorporated into numerous commercial products including desktop computers, laptops, PDAs, cell phones, keyboards, headsets and printers, with more products being constantly added to the list of Bluetooth enabled devices.
The Bluetooth subband codec (SBC) is a low computational complexity audio coding system designed to provide high quality audio at moderate bit rates to Bluetooth enabled devices. The Bluetooth SBC system utilizes a cosine modulated filterbank, for example, for analysis and synthesis. The filterbank may be configured for 4 subbands or 8 subbands, for example. The subband signals may be quantized using a dynamic bit allocation scheme and block adaptive pulse code modulation (PCM) quantization. The number of bits available and the number of bits used for quantization may vary, thereby making the overall bit-rate of the SBC system variable or adjustable. This is advantageous for use in wireless applications where the available wireless bandwidth for audio, and the maximum possible bit-rate may vary over time.
The Bluetooth community has developed various specifications that define how to use streaming audio over a Bluetooth link. This opens up Bluetooth technology to a whole new class of audio devices, such as wireless stereo headsets, wireless speakers, and wireless portable MP3 players. With the introduction of new Bluetooth specifications for streaming audio, new Bluetooth products such as wireless stereo headsets and wireless file streaming applications are becoming a reality. Wireless applications require solutions that are increasingly low power in order to extend battery life and provide a better end user experience. With existing systems, the computational requirements of high fidelity audio coding may make it cost prohibitive and challenging to add features such as streaming music to some wireless devices especially mobile devices.
Packet based telephony such as Internet Protocol (IP) telephony may provide an alternative to conventional circuit switched telephony, the latter of which may typically require the establishment of an end-to-end communication path prior to the transmission of information. In particular, IP telephony permits packetization, prioritization and/or simultaneous transmission of voice traffic and data without requiring the establishment of an end-to-end communication path. IP telephony systems may capitalize on voice over packet (VoP) technologies, which may provide a means by which voice, video and data traffic may be simultaneously transmitted across packet networks. The data may include video data.
Voice quality (VQ) is a metric, which may be used to define a qualitative and/or quantitative measure regarding the quality and/or condition of a received voice signal. Voice clarity may be an indicator of the quality or condition of a voice signal. Voice quality may be an important metric that may ultimately dictate a quality of service (QoS) offered by a network service provider. The following factors, for example, may affect the voice quality and/or condition of a voice signal—noise, echo, and delay or packet latency. However, the effects of these factors may be cumulative. In this regard, factors such as delay and latency may exacerbate the effects of echo, for example. Delays that may affect the voice quality may include, but are not limited to, routing, queuing and processing delays.
Various VoP specifications, recommendations and standards have been created to ensure interoperability between various network components, and to create an acceptable QoS which may include voice quality. For example, the International Telecommunications Union (ITU) ratified H.323 specification, which may define various processes by which voice, video and data may be transported over IP networks for use in VoIP networks. The H.323 specification addresses, for example, delay by providing a prioritization scheme in which delay sensitive traffic may be given processing priority over less delay sensitive traffic. For example, voice and video may be given priority over other forms of data traffic.
The H.323 specification also addresses voice quality by specifying the audio and video coders/decoders (CODECs) that may be utilized for processing a media stream. A CODEC may be a signal processor such as a digital signal processor (DSP) that may be adapted to convert an analog voice and/or video signal into a digital media stream and for converting a digital media stream into an analog voice and/or video signal. In this regard, a coder or encoder portion of the CODEC may convert an analog voice and/or video signal into a digital media stream. Additionally, a decoder portion of the CODEC may convert a digital media stream into an analog voice and/or video signal. Regarding the CODEC for audio signals, the H.323 specification may support recommendations such as ITU-T G.711, G.722, G.723.1, G.728 and G.729 recommendations. The ITU-T G.711 recommendations may support audio coding at 64 Kbps, G.722 may support audio coding at 64 Kbps, 56 kbps and 48 Kbps, G.723.1 may support audio coding at 5.3 Kbps and 6.3 Kbps, G.728 may support audio coding at 16 Kbps and G.729 may support audio coding at 8 Kbps.
The voice quality of a speech CODEC may be dependent on factors such as the type of encoding and/or decoding algorithm utilized by the CODEC. In general, some CODECs may utilize compression algorithms that remove redundant information from the analog signal. Such compression algorithms may permit at least a close replication of an original analog signal. In this case, the bandwidth required for transmitting any resultant signal may be reduced. Other CODECs may utilize algorithms that analyze the signal and retain only those portions that are deemed to be of cognitive importance. These algorithms may reproduce a close approximation to the original signal. Notwithstanding, in this latter case, bandwidth utilization may be superior to the former case where redundant information may be removed. Accordingly, depending on application requirements and hardware limitations, one or more algorithms may be utilized to optimize performance.
Moreover, although economic attractiveness of VoIP have lured network access providers and network transport providers away from traditional circuit switching networks, factors such as the extensiveness of embedded legacy systems and customer demands, for example, have dictated the coexistence of both packet switched and circuit switch networks. Accordingly, new technologies and techniques such as audio and video coding and decoding may be required to support various modes of operation utilized by each system.
Further limitations and disadvantages of conventional and traditional approaches will become apparent to one of skill in the art, through comparison of such systems with the present invention as set forth in the remainder of the present application with reference to the drawings.